asterisk users - Mar 2006

Friday March 31 2006
TimeRepliesSubject
11:52PM 0 asterisk-stat and webmeetme by areski
9:40PM 0 testing list mail - please ignore
6:36PM 2 Zap channels - help
4:56PM 0 Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
4:49PM 0 Re: Asterisk-Users Digest, Vol 20, Issue 226
3:21PM 2 Iaxmodem speed limit?
1:55PM 23 1.2.6 doesn't use mpg123?
1:17PM 3 Display Name
1:07PM 1 Play wav while in connection with a caller
12:54PM 0 OT: ad-hoc polycom network
12:08PM 0 hosted billing service
11:47AM 0 MeetAsterisk Europe:: Get an Asterisk one-day introduction!
11:03AM 0 chan_nbs and nbs
10:57AM 6 Dial from php
10:53AM 0 Transcoding on asterisk
10:42AM 1 incoming triggers seperate outbound
10:29AM 3 Asterisk Referral - Cleanup on Aisle 7
10:24AM 8 Confused on Agents and Queues
9:33AM 3 PRI issues
9:06AM 5 Echo cancellation problem
8:58AM 1 transcoding g723 or g729 on asterisk
8:56AM 1 statechange_queue
8:11AM 0 decrease the speed of reading text!!!
8:08AM 4 cannot set outgoing cid
7:57AM 6 IAXY codec support and questions..
7:55AM 0 meetme option 'e'
7:47AM 2 Asterisk hosted solution
7:15AM 1 oh323 - unable to install
7:01AM 13 Building Asterisk embedded device
7:00AM 2 I have debug off why are the logs show debug info
6:05AM 1 Echo still present with Eicon Diva Server 4 Bri
5:53AM 8 IAX: Auto-congesting call due to slow response
5:16AM 1 Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.
5:03AM 4 Howto cut the first digit
4:30AM 1 Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
4:24AM 0 How do you perform a Variable Substitution In Asterisk
4:01AM 6 Asterisk, QSIG and Tenovis PBX?
3:48AM 3 Quintum Tenor DX4060
3:24AM 1 Outgoing SIP Failover
3:07AM 0 No voice heard in festivalassociated with asterisk!!!
2:41AM 3 asterisk turn key solution
2:29AM 0 bristuff does not work with TDM400P
1:09AM 4 How to check if a phone / line is used?
 
Thursday March 30 2006
TimeRepliesSubject
11:38PM 1 Multicast Music on Hold
9:46PM 1 Using Voicemail with MP3 files...
9:17PM 1 DID billing
8:16PM 8 Anybody know about Cisco VOIP routers?
7:08PM 13 caller anounce
5:24PM 1 Questions on call recording and conference.
4:49PM 0 SIP Plugin or Component
3:54PM 4 Config TE110P and TDM400 with 2 FXS modules
3:07PM 1 Zaptel compile errors
3:03PM 4 multiple auto attendants
2:39PM 13 Asterisk and Hylafax, on the same box
2:33PM 6 OT: Polycom IP501 and Speed Dials
2:23PM 0 Vmail.cgi and #include
2:19PM 1 misdn timeout?
2:13PM 6 Please Help Test Quad PRI Using NFAS
1:28PM 2 Authenticate
1:22PM 0 PRI channel hangs after BUSY
1:11PM 0 DID's Now Offering Romania Bucharest 4021+ and 4031+
1:08PM 0 How we tell who is using VAD ?
1:05PM 3 Disable polycom call waiting?
1:02PM 0 BUG: FOP reports incorrect (duplicate) IP address until restarted
12:38PM 7 Reload astdb?
11:13AM 1 'sip show users' shows NAT RFC3581
11:11AM 1 sending SIP text messages to capable phones from an app
10:28AM 0 AMP backup-restore problem
9:58AM 1 internals and ISDN calls fail when Internet is down
9:34AM 0 Wrong extension indicated when logging in as agent
9:28AM 1 Benchmarking an Asterisk Server with 14k users
9:24AM 4 Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
9:15AM 0 SIP: INFO before answer causes disconnect
9:14AM 29 How is Teliax ?
9:02AM 2 Connecting a Grandstream Handytone 486 to Asterisk
8:56AM 0 Setting up announcement on reply to 4xx 5xx 6xx messages
8:42AM 5 Span monitoring
7:46AM 0 Strange second REINVITE being sent
7:20AM 37 Asterisk in production as a fax server, anyone?
7:01AM 3 asterisk doesn't wait for whole extension
6:18AM 3 Callid on T-1 trunk
4:07AM 1 Asterisk out of Media Path - Call Park
3:32AM 3 TDM04B sound volume
3:27AM 2 Panasonic KXTD 1232 6
1:43AM 0 Why would asterisk presume a loop (482 "Loop Detected")?
12:53AM 0 mixmonitor and how to stop it
12:45AM 0 Any new Voice Recognition devs?
12:28AM 0 Re: Asterisk-Users Digest, Vol 20, Issue 211
 
Wednesday March 29 2006
TimeRepliesSubject
11:58PM 0 R: RE : Echo cancellation
11:56PM 1 SV: IAX - only one way traffic
10:40PM 6 Problem with setting ringtones on Cisco 7960 phone.
10:01PM 3 Realtime Users/Peers/Friends - Ick
8:01PM 0 src/chan_h323.c: Failed to initialize OOH323
6:11PM 3 Blacklist out bound numbers from file
4:52PM 1 OT: FOP and reverse_transfer
4:52PM 2 Asterisk as Voicemail Server for Option 61c?
4:34PM 0 eyeBeam v1.1
3:38PM 7 Asterisk Between PBX and FXS
3:37PM 4 Dumb question - reaching the PSTN
2:02PM 0 Streaming voice using IAX
1:51PM 0 Two X100p clones. One not available for outbound?
12:58PM 2 Unable to open Asterisk database
12:19PM 9 Regulatory Ruling about Caller-ID
12:12PM 11 Inter-Asterisk Using SIP
11:29AM 3 H323 behind a Firewall
10:59AM 2 Calling home while on the road, will it work?
10:35AM 6 zaphfc on an 'actual' asterisk?
10:08AM 1 OT: HOWTO: Query channel state on an Ateus Voice Blue GSM gateway
9:42AM 0 indications.conf.sample
9:34AM 1 SJphone Do not send silence - option ? Should be disabled for Asterisk
9:32AM 0 Installing Cisco IP phone 7910
9:26AM 2 cdr_odbc appears to have fields missing
9:02AM 6 AAH lost my IVR phrases
9:01AM 8 FOP flash panel: how to reload config files when running
8:45AM 5 IAX - only one way traffic
8:35AM 0 inbound routing help
8:08AM 7 Routing SIP calls via URI
7:37AM 11 Marketing Materials
6:59AM 15 Asterisk with Vonage
6:55AM 9 SMS in Spain (it seems Protocol 2)
6:51AM 1 Oneway Audio
6:05AM 8 Reporting?
2:06AM 1 Avoiding initial deadlock on iax?
 
Tuesday March 28 2006
TimeRepliesSubject
11:12PM 0 Realtime mapping problem after svn upgrade
7:59PM 4 Call Monitoring / Call Takeover with Asterisk
6:21PM 0 IAX2 errors
6:05PM 6 dial plan logic
5:22PM 1 Dialogic d/4 PCI
4:12PM 0 H323 Info
4:04PM 1 Asterisk Tools for OSX
4:03PM 2 Transferring calls - BUG0003710
3:58PM 4 RTP frame size location?
3:31PM 0 Can realtime extensions be used within AEL contexts?
3:22PM 1 Asterisk to MySQL Data Lookup Warning Message?
2:47PM 1 RXgain
1:12PM 2 Problems Configuring Cisco 12SP+
1:08PM 2 Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels?
1:05PM 1 Redirect problem/bug/feature
1:05PM 1 AAH Mailing list
12:57PM 3 How to send announcement after called has picked up the phone?
12:26PM 3 Softphone accepting URL
12:07PM 1 Asterisk eating CPU
11:48AM 0 Bluetooth headsetin handsfree modewith SJPhoneorX-lite
10:40AM 4 Asterisk 1.2.6, VMWare, & Playback/Background GSM prompts
10:39AM 0 WARNINGS For SIP call
10:33AM 1 Monitoring question
10:21AM 0 R: R: Echo cancellation
10:05AM 4 Psgw
10:01AM 0 CID passthrough
9:44AM 3 Unable to authenticate password - VM
8:41AM 0 Bluetooth stack for cordless telephone
8:29AM 2 NATted phones transferring calls - BUG0003710
7:53AM 4 R: Echo cancellation
7:51AM 2 Agents on DND still receiving calls...
7:45AM 3 aah 2.7 / BRI
7:32AM 1 Asterisk 1.2.6 on Solaris 8?
7:27AM 0 h323 channel driver for production
7:25AM 0 codec translation problem???
7:13AM 9 Agent in multiple queues?
5:28AM 2 Squished faxes with txfax
5:19AM 8 Set caller ID for outgoing PRI calls
5:15AM 0 Injecting faxes directly to queue?
4:49AM 0 DTMF recognition inconsistent in Asterisk
4:48AM 2 IAX problems - please help me
4:31AM 6 Problems with wcte11xp module
3:05AM 0 Addons 1.2.1 upgrade to 1.2.2
2:47AM 8 Dial out .call files File permissions??
2:38AM 6 ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
12:16AM 3 time update (7905)
 
Monday March 27 2006
TimeRepliesSubject
11:35PM 0 how to dial to hardphone by kiax
8:39PM 0 TDM11B desperate Help wanted
8:11PM 1 restart problem
5:38PM 0 BUG 0003710 - RE: Transfer Calls - REFER
5:19PM 6 Dell 2850 w/TDM2400?
4:42PM 4 definity g3 voicemail
4:40PM 0 Transfer Calls - REFER
4:27PM 0 SER inside of Asterisk is SCALABLE ?
4:17PM 1 Bluetooth headset in handsfree modewith SJPhoneorX-lite
3:51PM 0 ANNOUNCE: WIST - Web Interface for SIP Trace
3:45PM 0 oh323 signal update support
3:31PM 3 How to disable event_log?
3:20PM 8 sipura spa2 + asterisk bug ?
3:14PM 2 TDM400P busy
2:36PM 0 Intel Compilation Questions - Asterisk 1.2.5/6
1:29PM 2 FXO without answer supervision
12:59PM 1 Bluetooth headset in handsfree modewith SJPhoneor X-lite
12:55PM 2 queue caveats
12:43PM 0 TE 205P/A102 fit in hp dc7600?
12:25PM 4 Master.csv Shell Script
12:18PM 2 Call Waiting Issues
12:09PM 0 Wanted: Cd-bootable Fedora+Asterisk
12:06PM 10 Alarm on Unicall
11:51AM 1 Asterisk 1.2.6 and Zaptel 1.2.5 Released
11:32AM 20 Receptionist Phones (was 3Com Phones)
11:23AM 2 Searchable forums
11:20AM 2 Ability to put call on hold via manager?
10:46AM 0 Unicall Question
10:15AM 10 FreePBX & AAH
9:55AM 0 SIP caller id
9:48AM 1 Polycoms and hints
9:36AM 7 CLI Echo
9:01AM 3 Testing asterisk faxing functionality
8:57AM 0 Inaudible voice and sleepy voice
8:48AM 4 Config File Management
8:37AM 2 after-queues
8:19AM 1 Small - Medium Billing Software needed
8:04AM 1 Bluetooth headset in handsfree mode with SJPhoneor X-lite
7:21AM 0 Bluetooth headset in handsfree mode with SJPhone or X-lite
7:05AM 1 FW: Re: Fw: anybody has SIP realtime working ?
6:41AM 0 Timeout waiting for response to Originate
5:46AM 5 automatic callback when busy
5:27AM 0 Who hangup.
5:21AM 0 Question about Polycom 601 and expansion module.
4:56AM 3 Voicemail to Email
4:51AM 0 Re: Re: Cisco 7960 - Have to press a menu button to dial
4:47AM 0 Transfer after group pick-up
4:38AM 2 Call Simulator
4:24AM 3 registration with different username
3:57AM 5 Polycom 501 Output volume
3:08AM 2 Caller ID length
1:57AM 3 Any Polycom dealer willing to help?
1:49AM 0 get no connection, very often, but not allways, why?
1:35AM 0 iax2_poke_noanswer on IP change. Sometimes permanent.
1:34AM 4 Alarmreciver
12:08AM 1 7940 with Asterisk?
 
Sunday March 26 2006
TimeRepliesSubject
11:22PM 1 Re: Cisco 7960 - Have to press a menu button to dial
11:22PM 2 Free g729
11:22PM 0 Asterisk add-ons upgrade
7:06PM 2 Web based voicemail client
5:57PM 4 Snom 360 - Multiple Server BLF Indications
5:41PM 0 RE: Hopefully a Simple Question?
4:56PM 0 Jittery Linksys/Sipura meetme conference fixed
4:07PM 0 UK EI
3:03PM 0 SIP realtime: how to authenticate without "name" field ?
2:57PM 0 MusicOnHold with mpg123
11:00AM 0 RE: Asterisk-Users Digest, Vol 20, Issue 184
10:44AM 1 AAH: DNID not set if caller suppresses CID?
10:30AM 0 hang up when pickup analog phone
9:19AM 3 tsu-600
8:20AM 0 Polarity reversals on a TE100P
6:14AM 0 zapata configuration & parsing
3:28AM 1 What codec extensions using now?
 
Saturday March 25 2006
TimeRepliesSubject
11:28PM 2 Copying SIP Subscriptions
11:06PM 3 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing
9:39PM 6 compiling Zaptel-1.2.4 on CentOS 4.3
5:28PM 0 Mailing list problems with gmail!!!!
5:23PM 0 CLI notice: channel.c:2421 __ast_request_and_dial: Don't know what to do with control frame 15
3:59PM 6 Asterisk and "Commercial Unix"
3:15PM 5 Disable timeout for answered queue calls?
3:06PM 2 On site installtion Tech. wanted
7:55AM 5 Asterisk billing from CDR database
5:23AM 3 QuesCom 400 IP/GSM
4:56AM 2 Asterisk spanDSP / Faxing problem
4:45AM 0 WARNING[3675] samples/codec_g723.c: Received a G.723.1 frame that was 4 bytes from RTP
2:45AM 0 apic vs xt-pic on fc3?
2:36AM 12 help on mfc/r2
1:22AM 2 Error in starting * with latest trunk
1:06AM 16 Polycom IP 301 is slow
 
Friday March 24 2006
TimeRepliesSubject
10:53PM 28 IAX Incoming/Outgoing
9:24PM 0 Bluetooth headset with SJPhone
7:03PM 3 iax limit question
6:21PM 11 GSM/DECT handsets (was gsm picocells)
6:10PM 7 Polycom 601 Message Center
5:47PM 2 [1.2.5] DTMF not being set correctly (RESEND)
5:19PM 11 3Com Phones
3:28PM 1 Re: Subscription state after reload (New subject)
3:11PM 2 SV: re: Sound issues on SIP-SIP calls
2:51PM 1 PRI Behavior
2:19PM 4 SIP trunk problem
1:17PM 0 Finding the busypattern
12:40PM 21 Transferring a call with IAX
12:01PM 1 making ooh323 authenticate gateway just like sip does
11:42AM 0 FW: Extension a?
11:24AM 0 Queue Period-Announce
11:18AM 0 Realtime Agents
11:11AM 0 VoIP QoS monitoring and failover re-routing
10:55AM 18 Asterisk Failover without SER
10:47AM 1 Maximum Queue Name Length
10:42AM 5 FW: Asterisk Users
10:41AM 10 Snom 360 problems
10:23AM 0 Echo and static when dialing Asterisk
10:03AM 6 Best GUI for basic HostedPBX service
9:10AM 2 Getting True ANI not Caller ID
8:25AM 3 Plain Old Answering Machine
8:05AM 5 * Meetme Freeze patch found
7:43AM 0 On ParkAndAnnounce and parking lot
6:37AM 5 Mandrake zaptel module not found after compiling
6:25AM 0 Hints in Realtime
5:11AM 0 RES: reload - restart
4:49AM 1 reload - restart
4:01AM 3 Call terminated after 60 seconds
3:35AM 0 Which 2 Port ISDN Card for P2P (Austria)
3:00AM 1 Re: Server freeze with meetme and sip GSM users
2:50AM 2 How to nice agi scripts?
2:48AM 0 Speed up dial using #?
2:21AM 0 UK pri almost working
2:07AM 2 Problem with MeetMe Conference!!!
1:27AM 0 pots -> asterisk -> tsu-600
12:42AM 3 chan_h323 problem
12:36AM 3 getting your own phone number
12:09AM 1 problems compiling zaptel on FC5
 
Thursday March 23 2006
TimeRepliesSubject
11:59PM 0 CallerID chopped by half ? :-)
9:57PM 0 What do the Queue timeouts really mean?
9:29PM 4 Page about 70 users crash my Asterisk
9:25PM 7 Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)
8:44PM 0 GnuGk and Asterisk IVR
8:26PM 0 realtime and queues and persistantmembers in 1.2.5
7:57PM 2 IAX Bridging and not recording CDR correctly
7:17PM 11 FXS channel banks
6:11PM 3 Which g729 codec to download for a P4?
6:00PM 13 Tearing my hair out with Queues
5:39PM 6 Polycom 501's for sale
4:48PM 6 Which Mac OSX softphone with IAX2 support?
3:50PM 13 [OT] Polycom provisioning
3:35PM 0 7940/60 SIP Call Park Button
3:34PM 0 Re: Subscription state after reload (New subject)
3:29PM 5 kernel recompilation on a asterisk server
3:28PM 0 Re: Subscription state after reload (New subject)
3:21PM 1 transfer incoming call to VM without answering call
3:01PM 0 Anonymous sip calls getting into wrong context?
2:57PM 1 wellgate 38XX FX & FXS voip gateways with outgoing call files
2:45PM 1 User Extension Custom Voicemail
2:32PM 2 Dialling Problem
2:25PM 2 Changing codec.
1:42PM 2 TAC Case Cisco 7960 Proxy address showing up in callerID
1:32PM 4 Aastra 9331i phones
12:40PM 0 Cracked and sleepy voice
12:18PM 18 How to create [new_context] in extensions.conf?
12:05PM 2 RE: MeetMe freezes machine with Junghanns
12:05PM 9 I'm FED UP with BroadVoice
11:54AM 1 High Density Analog
11:25AM 5 MeetMe freezes machine with Junghanns QuadBRI cards
11:16AM 10 Ok... what is 'sip show peers' really used for?
10:57AM 1 Call Monitoring?
10:53AM 4 Call Recording?
10:24AM 1 Cisco 7970 SIP Image - hint lines
10:19AM 2 "Not Found" in archive
9:54AM 0 Problem with INVITE's being sent
9:52AM 0 Arcavox / AV-Global
9:51AM 1 Problem with Queue periodic announcemnets
9:00AM 2 SIP - Problem with audio clipping
8:41AM 0 How to monitor dropped frames on a PCI bus
8:38AM 2 Maximum retries exceeded on transmission
8:31AM 0 Billing from CDR files
8:10AM 0 Zaptel compilation problem on SUSE
8:09AM 0 compiling chan_h323
7:55AM 0 Sending 2 CallingNumbers
7:28AM 0 Trouble installing a digium TE405P card
5:50AM 3 best MTU?
4:34AM 1 CHINA DID
4:05AM 2 asterisk as a fax server
2:29AM 2 Netgear FS116P and Cisco 79XX phones
1:21AM 2 type of incoming lines
 
Wednesday March 22 2006
TimeRepliesSubject
11:24PM 11 welltech Wellgate 3804 in SIP mode
10:54PM 0 Video phone failed on Asterisk-1.2.4
9:40PM 0 AstriCon Europe: Early Bird Open / Speakers & Papers Wanted
9:25PM 3 G729 License questions
8:16PM 1 Hardware Levels
7:41PM 2 polycom queue bug
7:21PM 1 Polycom IP501 Buddy List
7:17PM 6 7970 8.x firmware speeddials
5:49PM 2 Re: Asterisk-Users Digest, Vol 20, Issue 153
4:21PM 4 PRI DMS100 -> Nortel Meridian Option 81
4:01PM 2 connecting Avaya Partnet with asterisk , TE205P
3:52PM 10 Asterisk Users
2:58PM 1 pseudo Direct Outward Dial
2:50PM 5 Stability and motherboard questions with TE406P and TE410P
2:45PM 4 best CENTOS to use for latest asterisk
12:50PM 10 Voicemail limit?
11:57AM 7 Can this box handle 8 T1s (PSTN) with Asterisk?
11:05AM 15 VERY IMPORTANT(TREAT WITH URGENCY)
10:47AM 1 Big Traffic anyway?
9:55AM 4 Realtime Query
9:39AM 0 call drops after one ring
9:24AM 3 router UDP timeout
9:00AM 1 Dial plan question - exclamtion mark
8:54AM 2 Asterisk snapshots?
8:43AM 0 SPA-2002 Upgrade Question
8:38AM 2 Asterisk--->>Autodialling
8:34AM 3 what are these and can they be fixed?
8:31AM 2 License for asterisk-addons?
8:28AM 0 TIMEOUT(s)
8:10AM 6 Remote dialtone
7:47AM 0 the best configuration for DTMF detection on SPA 2000
7:18AM 0 RE: VoiceMailMain(@context) Problem with Opt ion 5 (Advanced)
7:07AM 0 ZOMBIE on att transfer
6:31AM 5 Sound issues on SIP-SIP calls
6:27AM 5 Double Call Progress tones
5:04AM 2 beronet & bristuff
4:50AM 1 How to hide CallerID - SetCallerPres(prohib) not working
4:44AM 3 Asterisk & Avaya Legend
3:54AM 2 Asterisk perms in manager.conf
3:09AM 0 Help! Directing Inbound calls to different extensions
1:34AM 2 Pickupexten not working
1:01AM 5 transfer calls via Manager Api
 
Tuesday March 21 2006
TimeRepliesSubject
11:53PM 5 Asterisk and gateway
11:16PM 6 PSTN to Asterisk VOIP in Manila
10:05PM 0 PRI not answering call after asterisk upgrade
9:51PM 8 Programming the Manager API
9:44PM 2 Multiple commands per priority
9:35PM 1 SIP video voicemail problem
9:24PM 7 TDM400 FXO module not answering or dialing out.
9:02PM 0 IAX2 issue with calls transfered between systems.
8:32PM 1 Cannot leave voicemail, Asterisk/Zaptel/libpi v1.0.9
5:01PM 2 'Click to Dial'
4:10PM 0 Modem Bank
3:57PM 0 [OT] Cisco 7970 SCCP Image
3:44PM 11 ADT/Brinks alarm dialing through Asterisk
3:35PM 1 Polycom hand/head set echo and Zapata config
3:17PM 1 VoiceMailMain(@context) Problem with Option 5(Advanced)
3:06PM 0 Attachment Blocking Notification
2:52PM 2 VoiceMailMain(@context) Problem with Option 5 (Advanced)
2:51PM 49 Fw: anybody has SIP realtime working ?
2:37PM 1 Dimensions of TDM 24 card
1:57PM 16 FAX over PRI
1:36PM 4 callerid= in zapata.conf
1:09PM 0 Blind Transfer / Ring Groups Problem
1:05PM 8 Remote MWI over IAX2
12:53PM 0 Sound dies
11:31AM 5 Realtime SIP Persistency
11:24AM 2 need to make my oh323 work with quintum no gatekeeper
11:23AM 0 Asterisk with TOPEX GSM Gateway
11:19AM 3 Problem with chan_iax.cimplimentationcausesbadaudio?
11:07AM 6 Realtime / SIP Peers etc
11:00AM 0 DTMF leak with IAXy & call waiting Bug?
10:56AM 1 ODBC and VoiceMail messages.
10:48AM 4 VoIP prepaid billing
10:21AM 0 SIP Realtime 1.2.5 and Username/auth name mismatch ?
10:07AM 5 Cisco POS 3-08-2
10:02AM 4 WiFi phones and WDS (Wireless Distribution System)
10:02AM 17 Native MOH - Convert mp3 to ulaw
9:44AM 0 hfc-pci cards on ppc
9:37AM 8 Multiple processes
9:19AM 3 Problem with chan_iax.c implimentationcausesbadaudio?
9:16AM 5 Junghanns and Digium TDM400?
8:55AM 1 Problem with chan_iax.c implimentation causesbadaudio?
8:51AM 6 Zap<-->IAX codec?
7:48AM 1 Caller ID forwarding with Pickup() application?
7:47AM 1 Problem with chan_iax.c implimentation causesbad audio?
7:46AM 0 app_queue and ARA
7:13AM 2 Voice mail not working with Asteriks 1.2.5
7:09AM 2 Web-ex type solution for use with asterisk
2:53AM 21 How to make groups of extensions ???
2:50AM 2 How to make extension groups ???
2:00AM 0 Queue and busy/congested ZAP channels
1:34AM 0 CDR problem with TAPI
 
Monday March 20 2006
TimeRepliesSubject
4:47PM 12 Problem with chan_iax.c implimentation causes bad audio?
4:36PM 0 problems with international dialing
3:29PM 1 Asterisk Disconnecting after 30sec when someone leaving VM
2:28PM 0 integration with Toshiba PBX system
1:08PM 0 Primary D-Channel on span 1 down
12:57PM 0 Experiences with ATA model Octtel SP200SO
12:28PM 5 Problem with intermittent one-way audio
11:28AM 0 sip show inuse not accurate
11:18AM 1 (no subject)
11:07AM 1 Is it possible to turn off password for transfers on FOP
10:44AM 8 Aterisk with Realtime
10:04AM 0 meetme recording very loud
9:35AM 0 assman, the ncurses asterisk manager interface
9:21AM 1 simple question on asterisk
9:19AM 3 pickup a call in queue
9:14AM 0 Prodding channel h323
8:52AM 3 How often do YOU register?
5:54AM 17 answer delay
5:24AM 2 How to make caller groups ???
5:12AM 0 How to setup Proxy info to * box , [* box behind a squid proxy and firewall ]
4:53AM 0 MixMonitor and transferred calls
4:12AM 0 Help: Using asterisk and mysql for a university project
3:56AM 8 simple perl-agi - where's the error?
3:51AM 5 ISDN Protocol Unknom Error with Junghanns OctoBRI
3:14AM 0 Problems loading res_odbc.so and cdr_odbc.so
2:38AM 3 Grabbing the billsec and duration after a hangup.
2:32AM 6 Numbered Voicemails even with delete option!
1:32AM 0 Using IAX
1:16AM 0 asterisk and DDI
12:32AM 6 计划生育的无耻宣传该结束了
 
Sunday March 19 2006
TimeRepliesSubject
11:41PM 0 HFC USB (was MultiBRI in Australia - found one-maybe)
9:42PM 2 Local Channel
8:30PM 0 Xorcom TS-1 T1 installs?
4:40PM 1 HFC USB (was MultiBRI in Australia - found one - maybe)
4:02PM 0 Transfer to specific park number
3:24PM 16 Call Pickup Woes
3:24PM 0 ISDN NT Mode & CAPI
3:04PM 2 Grandstream unit HT-488
2:40PM 0 Problem w/ Dial Command on Zap channel
2:40PM 1 trunking questions
2:32PM 0 Calls to SIP providers
12:40PM 3 Zaptel will not build
11:39AM 0 Voicemail Bug?
10:57AM 0 Bizzare DTMF on channel bank
10:56AM 3 Annoying Asterisk Realtime Limitation
10:04AM 0 Sending ANI to TDM40B FXS?
7:41AM 1 accessing speed dial database
1:48AM 0 PSTN lines permission settings to different extensions
12:31AM 17 An FXO version of IAXy?
12:20AM 6 g729 and latency measures
 
Saturday March 18 2006
TimeRepliesSubject
8:32PM 2 A general deployment question (OT)
7:51PM 2 Polycom IP600 - no ring?
7:09PM 0 Polycom IP600 dual ethernet port - bandwidth impact
6:42PM 2 GS BT102 dual ethernet port -bandwidth impact
6:22PM 0 Cisco 7960 dual ethernet port - bandwidth impact
3:52PM 0 Panasonic KX-TDA1000 with asterisk server
12:21PM 1 Realtime SIP users/peers - Screwed?
12:16PM 2 Jittery meetme conference using Linksys 942 phones
11:54AM 0 Realtime SIP users/peers
11:09AM 4 How to enable talking in chanspy while spying?
9:42AM 4 Sipura 3000 DMTF
9:02AM 0 I have my asterisk machine behind a Linux, Nat ...
8:12AM 1 List of transcoding combinations
2:01AM 0 Re: Server freeze with meetme and sip GSM users
12:43AM 0 I have my asterisk machine behind a Linux Nat ...
12:03AM 0 T38 Passthrough testing -- unknown media type error
 
Friday March 17 2006
TimeRepliesSubject
10:57PM 0 Question on compiling Zaptel
10:54PM 1 Re: DUNDi .... Halfway and CLUSTERING
8:53PM 0 Critical Problem with asterisk
8:33PM 47 Asterisk Users Mailing List Traffic
6:11PM 19 gsm picocells
4:34PM 4 SIP Realtime Users
4:16PM 0 New astGUIclient VICIDIAL Released: 1.1.10
3:21PM 1 One-Way SIP Audio with SVN Codebase (CANCEL)
2:43PM 0 OT: Good Vendor?
2:29PM 0 FreePBX 2.0.1 released!
2:23PM 3 Exchange 12 Unified Messaging
2:12PM 1 Re: DUNDi .... Halfway and CLUSTERING
1:34PM 1 More Voicemail prompts
1:15PM 7 Re: DUNDi .... Halfway and CLUSTERING
12:45PM 1 how to get separate CDR for inbound and outbound legs of a call
12:40PM 0 Call transfer problems, SOLVED
12:39PM 1 Extra Debugging without console
12:37PM 4 IAX Phone?
12:13PM 0 How to set priority for SJPhone.
10:33AM 1 problem with tdm22b
10:32AM 0 Transfer problems revisited
10:11AM 8 Aastra Questions
9:47AM 6 Disappearing voicemail
9:38AM 0 caller unable to transfer
9:08AM 1 Sticky Problem SER/Asterisk
9:00AM 0 CRM + Phones
8:55AM 5 Countries supporting SMS on PSTN (ISDN)
8:16AM 0 Asterisk and PacketCable
8:05AM 24 problems with emailing voicemail
7:55AM 2 DISA alternative
7:36AM 1 RE: DUNDi .... Halfway and CLUSTERING
7:13AM 2 choppy recorded sounds in asterisk
7:10AM 1 Asterilink?!?!
6:59AM 0 (no subject)
6:47AM 4 TDM 2400 With 24 FXO
6:46AM 4 D4 AMI - No Caller ID
6:42AM 2 Analog POTS line -> Rhino FXO Channel Bank -> No Hangup
6:31AM 0 Set CallerID to a specific Queue Member
6:27AM 0 [FOLLOWUP]: Calls not tearing down properly
6:19AM 0 Keeping the user name in sip INVITE with fixed IP host routing.
6:07AM 1 Asterisk on hosted server
6:03AM 0 How to install the cdr_odbc module.
5:11AM 6 Best budget IP phone at the moment?
4:55AM 0 Call pickup between different protocols
4:15AM 0 One-Way SIP Audio with SVN Codebase
4:12AM 2 asterisk and skype - asterisk newbie
3:03AM 4 TFTP problems on FC4
2:45AM 3 embedded hardware for Asterisk?
2:13AM 1 automatic fax detection in asteriskathome
1:49AM 0 OT: reset LinkSys 941 to factory defaults & howto config' via TFTP
1:48AM 3 Numbered Voicemails when you still delete them.
1:43AM 0 asterisk configurations
1:34AM 0 Echo/Milliwatt Test Numbers in Oz ?
1:29AM 0 OT: any one in Stockholm for quick piece of advice
1:16AM 1 french sounds in asterisk
 
Thursday March 16 2006
TimeRepliesSubject
10:30PM 1 Feedback from VON expo!Infoon*HAandPolycomphone!!
10:27PM 0 RE: DUNDi .... Halfway and CLUSTERING
10:25PM 3 Question about advanced IVR
10:23PM 0 qozap drops -- possible to bridge BRIstuff ISDN to analog zaptel phone?
8:50PM 0 Feedback from VON expo! Infoon*HAandPolycomphone!!
7:38PM 18 OT: Unblocking bloced CID
5:25PM 0 Creating Asterisk Bounties
5:24PM 1 DUNDi .... Halfway
5:17PM 0 Small noise every 3 seconds
4:36PM 3 Queues Not Reporting Estimated Hold Time
4:28PM 1 UK Caller ID - Asterisk 1.2.5 - TDM4 Card
4:28PM 5 New one on me: How to UN-transfer
4:19PM 0 redirect output
3:58PM 0 (no subject)
3:48PM 1 Newbie needs audio help
3:37PM 0 Implementing VoIP for first time with Packetcable
3:30PM 5 voip-info.... again
3:24PM 0 SCCP problem with ATA188, Asterisk@home and chan_sccp
3:14PM 0 Two dissimilar data only T1 lines possible in Asterisk?
3:10PM 0 Regcontext, only 1 context available?
2:46PM 2 Authenticate CDR Logging
2:38PM 1 G.729 codec licencing
2:35PM 0 3 way calls & transfers
1:42PM 0 Asterisk select outbound trunk based on minutes used per month??
1:36PM 1 RFC 2833 and SIP? DTMF? What am I not getting?
1:34PM 1 MFCR2
1:26PM 1 Re: transfers/parked calls + polycom 501
12:03PM 4 Feedback from VON expo! Info on * HA and Polycomphone!!
11:54AM 2 Can anybody get me setup with a hosted asterisk@home box or virtual server in the next 24 hours?
11:40AM 6 Re: transfers/parked calls + polycom 501
11:36AM 0 Budgetone strange problem - have to press hold on and off to connect call.
10:45AM 2 capiHOLD missing in BRIstuff 0.3.0
10:31AM 0 Dialplan : Forwarding call to voicemail after onering iif extension is busy
9:45AM 0 Dialplan : Forwarding call to voicemail after one ring iif extension is busy
9:35AM 1 MeetMe - Causes * to crash :/
9:35AM 1 Creating a voip network... use asterisk?
9:30AM 0 Zap channel not hanging up
9:29AM 0 Asterisk Users Group Tonight, Irvine, Ca
9:24AM 1 setting callerid not working if no callerid on incoming number
9:09AM 0 Testing IAX links
8:58AM 0 Fw: help required configuring card
8:55AM 1 Feedback from VON expo! Info on *HAandPolycomphone!!
8:45AM 0 Feedback from VON expo! Info on * HAandPolycomphone!!
8:30AM 3 Attended call transfer with GXP-2000
8:20AM 8 Feedback from VON expo! Info on * HA andPolycomphone!!
8:16AM 0 ODBC voicemail storage
8:09AM 3 module load order for Junghanns qozap and TDM card
7:52AM 2 ISDN BRI and UK Premium Rate Numbers
7:29AM 1 Queues - calls going to agents lised as "In use"
7:05AM 0 can't get TDM400P to answer
6:06AM 0 How to configure PSTN lines permissions todifferent extensions ???
5:11AM 3 open source queue analyzer
4:34AM 0 [Fwd: Re: Sync Source: Internally clocked]
4:24AM 1 Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)
4:04AM 11 asterisk@home V's Asterisk
3:02AM 0 Server freeze with meetme and sip GSM users and ztdummy
2:09AM 2 SER & Asterisk with DID incoming and out going
1:50AM 1 asteriskathome maximun channels per trunk
1:48AM 0 carry forward uniqueid
1:40AM 2 SIP routing over IAX2
1:05AM 4 How to transmit Video
1:04AM 0 Wanted: IAX ATA w/ FXO
12:55AM 1 send text to a device
12:07AM 2 Problem with System() command.
 
Wednesday March 15 2006
TimeRepliesSubject
10:11PM 2 How to configure PSTN lines permissions to different extensions ???
8:55PM 0 FXS Caller ID?
8:07PM 2 GUI Web interface
6:04PM 2 dropping voice frame ulaw - slin?
5:42PM 3 Do Not Disturb?
3:35PM 6 Failed to read gains: Invalid argument
3:07PM 1 Unable to forward frame
3:01PM 0 sporadic voicemail delete problems
2:46PM 6 Echo canceller data-points
2:26PM 2 Speeding up the dial of DTMF's in SIP channel
2:10PM 0 T.38 Passthrough testing -- IAX problem
2:04PM 0 RE: Asterisk-Users Digest, Vol 18, Issue 147
1:45PM 4 Sync Source: Internally clocked
1:45PM 0 (no subject)
1:12PM 0 (no subject)
1:06PM 2 Help with Gizmo from outside firewall
12:48PM 0 definity prologix
11:58AM 0 OT: Using Sipsak to reboot a Snom phone < -a nswered my own question
11:48AM 0 Call go on hold for no reason
11:46AM 0 Idiot's guide to Q.932?
11:40AM 0 OT: Using Sipsak to reboot a Snom phone
10:33AM 15 how to show called name on calling polycomdisplay
10:33AM 2 Script to Restart Zaptel
10:15AM 0 openSUSE 10.0 and zaptel init script
10:05AM 6 Double-ring tone
9:44AM 1 ooh323 Gatekeeper Bug
9:25AM 2 Fake Ring Tone/Compile Addon
9:21AM 0 res_config_mysql.so not found
8:50AM 2 Asterisk integration with office PBX
8:42AM 1 cisco 7912 not taking config
8:39AM 1 cards
8:37AM 13 OSHA requirement to "reach a live human" ??
8:17AM 0 Aastra 480i CT - multiple lines?
8:04AM 0 Re: Stuck. Extenions.conf? Realtime? MySQL?
8:01AM 4 Toshiba Strata DK-280 support?
7:58AM 1 Development news :: T38 passthrough
7:44AM 0 Attended transfers timing out after 3 rings
7:41AM 5 problem configuring a digium quad E1 card
7:39AM 0 Zaptel compile errors on x86_64 - DEFINE_SPINLOCK???
7:24AM 0 Meetme monitoring only bug
7:15AM 1 external modem
6:37AM 6 misdn problem
6:33AM 4 AVM C2 chan_capi-cm-0.6.3 Error on Dial
5:41AM 0 IMACS800
5:20AM 2 asterisk crash too much?
4:13AM 0 [SPAM] [asterisk-dev] CALL FOR COMMENTS - Dialplan
3:44AM 4 (unexplicable) peaks of machine load
3:23AM 0 spa 3000/2100 noise
3:23AM 0 there is lack behind in recoded calls via sox
3:19AM 0 There is lacking behind in recorded calls via sox
3:14AM 1 asterisk perl commands
2:52AM 9 Cisco phones and Linksys SRW224P
2:49AM 3 Zaptel compile errors on x86_64
2:29AM 0 MCC v.1.3 Released
2:08AM 3 How to assign a specific PSTN line to a specific extension ???
1:52AM 6 how to show called name on calling polycom display
1:37AM 1 IVR weirdness
12:32AM 5 Asterisk to receive fax
 
Tuesday March 14 2006
TimeRepliesSubject
11:16PM 1 invalid wav gsm frame size: 1 bytes ??
10:57PM 3 asterisk and iptables
10:53PM 1 RE: Problems with installing a TE110P on a Dell Poweredge 850
9:51PM 13 Asterisk Native Sounds - in case you missed it...
9:30PM 1 Bug Help or Suggestion - Grandstream GXP2000 (firmware 1.0.2.8) - BLF, Hints, call-limit
9:23PM 2 Adding entries on company directory
9:02PM 5 Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!
9:01PM 2 asterisk hang
8:01PM 2 isdn out of band signalling
6:41PM 1 Directory doesn't work well Asterisk@home2.7- try from PSTN with Digital recepcionist- Directory based on Last name
6:13PM 2 Max retries exceeded to host...
5:32PM 1 Problems with installing a TE110P on a Dell Poweredge 850 running Fedora Core 4
5:26PM 0 help required configuring card
4:16PM 11 New ncurses Asterisk Manager Interface
4:13PM 0 External transfer
3:24PM 0 ANNOUNCEMENT : A2Billing (Asterisk2Billing) - release v1.1
2:26PM 0 invoking a macro doesn't work
2:15PM 0 LNP / DID Service - Louisianna / Virginia
1:46PM 2 E911 from Remote Office via PRI
1:41PM 0 LCDPROC cient for Asterisk
1:34PM 0 ip telephony project
1:34PM 0 Flash on Unicall Channel
12:34PM 9 Realtime Extensions
11:50AM 3 Outbound paging dialplan example?
11:47AM 0 MWI & Asterisk Realtime Architecture
11:34AM 3 EICON Diva 4BRI
11:22AM 3 Voice volume using Monitor application
11:14AM 1 channel bridging
11:08AM 0 List Rules
10:52AM 6 OT - force Cisco phones to reboot
10:38AM 5 Attended Transfer - transfer timeout, how to change?
10:18AM 1 Codec Issue
10:10AM 0 Asterisk Users Group Meeting March 16, Irvine, Ca
9:44AM 15 IAX choppy sound
9:27AM 1 Bad FXS Module?
9:21AM 0 Asterish Guru needed in Phoenix ASAP
9:01AM 1 Re: Asterisk-Users Digest, Vol 20, Issue 91
8:48AM 4 digium.com redesign
8:31AM 0 Problem with uac_replace and corrupted From
8:12AM 2 Realtime SIP
7:57AM 0 Sample SER + Asterisk conf?
7:43AM 0 [OT?]SCCP image for cisco 7905g
7:08AM 0 Problem with poud key (#)
6:59AM 0 Line connections
6:34AM 1 10minutes to restart Asterisk@home 2.7
5:59AM 2 Latest Dell SC430 Compatibility With Wildcard
2:31AM 0 DATA CALLS annoying my system
2:19AM 0 Inbound sipgate number forwarding to differnet users
1:01AM 0 I can't resume a call on hold from zap device
 
Monday March 13 2006
TimeRepliesSubject
7:27PM 0 Call Parking Grandstream
7:13PM 0 Spam? Re: Unknown signalling method 'pri_cpe'
7:11PM 9 Clustering "NEW THREAD", Almost Working
6:47PM 7 Unknown signalling method 'pri_cpe'
6:32PM 2 CDR Bug?
5:32PM 5 Can One FXO Support Multiple Phone Lines?
5:14PM 5 Simple php script to monitor asterisk calls
5:13PM 4 DISA & SPA3000 issues
5:10PM 1 incoming limit, call_limit, or call-limit?
4:12PM 1 MWI to 7960's sometimes delayed or lost. Please advise.
3:32PM 0 (no subject)
3:27PM 4 slinear bandwidth
2:52PM 1 Seperate music on hold for SIP extensions
2:36PM 1 cisco 7912 ringlist.dat file format
2:31PM 0 Phase locked mode
2:20PM 0 Calls not tearing down properly
1:59PM 11 SIP Jitter Buffer for 1.2.5
1:42PM 1 All calls in queue go to agent that is down??
1:39PM 3 Dumb question (hang up detection/Zapata.conf)
1:14PM 9 Cisco 7960 8.2 callerID lists proxy?
12:47PM 2 Outgoing calls via Sipgate
12:22PM 0 Re: Regexten & Regcontext, working now
12:19PM 8 priorityjumping=no
12:18PM 0 chan_zap ast_pickup_call issue redux
11:28AM 1 Spam? Re: Failed installing zaptel
11:15AM 0 Asterisk end Festival
11:01AM 0 How to forward inbound sipgate calls to different users in my entreprise, (
10:33AM 3 Zaptel not compiling on lastest Centos 4.2 kernel.
10:31AM 0 Re: transfers/parked calls + polycom 501
10:27AM 0 Regexten & Regcontext
10:25AM 1 Considering Asterisk
10:22AM 6 Failed installing zaptel
10:20AM 1 Hardware timing source for MeetMe
9:58AM 0 channel manipulation
9:40AM 3 music on hold without mpg123
9:38AM 2 Diff between X100M and X100P?
9:37AM 2 Avaya IP Office 412
8:59AM 2 Cannot load wcfxo -- Please help!
8:47AM 0 Incoming Call keeps ringing when the second call arrives
8:35AM 1 misdn
8:20AM 8 Re: transfers/parked calls + polycom 501
8:06AM 2 Asterisk large scale, help needed
7:55AM 10 echo problem + choppy sound
7:55AM 1 Need help implementing call center featuresofAsterisk
6:49AM 1 Scrolling messages
5:48AM 1 Need help implementing call center features ofAsterisk
5:25AM 2 Need help implementing call center features of Asterisk
3:51AM 1 G729A
2:04AM 3 Callerid on transfer
12:52AM 10 Asterisk RealTime Question, Please help
12:19AM 1 Australian approved 4BRI PCI adapter preliminarytesting results
 
Sunday March 12 2006
TimeRepliesSubject
11:13PM 2 Action on phone pickup
10:54PM 9 Multiple IAX clients behind a firewall
10:21PM 27 stop monitor on transfer
10:05PM 0 Invitation
7:00PM 1 Australian approved 4BRI PCI adapter preliminary testing results
5:00PM 1 Flash zap trunk from Softphone or IP Handsets...
3:43PM 1 Hung IAX Channels
1:32PM 0 DTMF digits lost on TDM400 card
12:08PM 1 Looking for docs on adjusting txgain/rxgain
12:08PM 2 Understanding queue timeouts + possible bug found
11:27AM 1 Building a small Office EPABX with VoIP GW with Asterisk
8:17AM 1 interop problem: "Missing handling for mandatory IE 24 (cs0, Channel Identification)"
6:33AM 1 Calls from PSTN , answering, When transfered get Hungup 'Zap/1-1'
5:19AM 1 Call and then play IVR
4:58AM 8 Voice problem
2:31AM 1 Speakeasy VOIP + Asterisk?
 
Saturday March 11 2006
TimeRepliesSubject
8:57PM 0 how to check if ztdummy is working properly?
4:46PM 4 Polycom - directory dial
4:21PM 1 Autodial
2:13PM 0 Unicall and Fax detection
2:11PM 2 Limiting the number of concurrent calls for a group of SIP devices
12:53PM 1 OT: Flash/web site developer in Boca Raton FL required
12:37PM 0 I don't listen first seconds of audio from call - Asterisk integration with old PBX
9:23AM 0 Incompatible switchtypes
9:04AM 1 how to connect 3 or more servers via IAX ?
8:53AM 0 Clustering / Dundi
5:16AM 1 FW: I need to set NO CRC4 on zaptel.conf?
4:24AM 1 HITBSecConf2006 - Malaysia: Call for Papers
4:09AM 2 IVR dial by extension option..
1:26AM 1 hotel vmail and iax trouble
1:13AM 0 asterisk having problem in playing sounds
1:01AM 0 Odd CID issue calling SIP to SIP DID - anyone have this or can explain it?
 
Friday March 10 2006
TimeRepliesSubject
9:29PM 0 change voicemail folders
8:56PM 1 (no subject)
3:35PM 1 voicetronix and asterisk@home
3:28PM 0 FW: T1 card Ali
2:54PM 14 Development news :: T38 passthrough support
2:34PM 16 Analog Desktop Phone
2:25PM 1 IAX / Firefly handshake problem
2:08PM 4 RFC Follow Me Find Me script
2:07PM 3 Junghanns, Germany ISDN settings
2:01PM 5 Problem compiling zaptel on latest RHEL kernel (2.6.9-34.EL)
1:57PM 8 dipura 2002 auto dial or intercom
1:41PM 1 HOWTO initialize new kernel & kernel source without reboot
1:19PM 8 Asterisk programmer needed
1:14PM 0 Voice Mail woe
12:57PM 4 Menu in queue
12:22PM 53 Clustering
12:20PM 1 cidname via IAX2?
12:20PM 0 Voicetronix OpenSwitch / Sangoma Analog Card
11:04AM 1 Yet again: chan_zap.c: Unable to specify channel 4: No such device
10:52AM 5 7970 Configs
10:19AM 3 Disable flash transfers?
10:06AM 0 queue and service period
9:55AM 0 Background timeout and Read questions
9:45AM 3 ADPCM - vs - G.726
9:39AM 0 TDM400 DTMF Caller ID
9:29AM 5 Action after _caller_ has hungup(cmd Dial 'g'-option)
9:04AM 0 RE: Stable Hardware Combination Experiences
8:47AM 0 DUNDi Public and Private Key Question
8:32AM 3 ring (hunt?) group
7:57AM 3 Dial plans and forwarded phones
7:51AM 0 Forward from SER to asterisk can't hang up
7:46AM 0 pstn to asterisk, DVG-3004S, MP104?
7:29AM 0 Operator consoles for large systems
7:09AM 0 Sangoma A101 T1/E1 (PRI) voip card available for testing
7:04AM 7 difference between records in CDR and real duration of call
6:15AM 9 IAX2 + Sonicwall
5:48AM 2 ZT_CHANCONFIG failed on channel 2: , Guidance requested
5:25AM 0 Flash call transfer problem
4:22AM 4 Dial Out IVR
3:20AM 0 ALSA channel (console/dsp) problem
2:59AM 1 monitor/statistic web interface for cdr
2:00AM 1 Can I avoid configuring FXS part in zaptel.conf and zapata.conf
1:50AM 0 mysql asterik
1:49AM 2 7960 Cisco SIP Phone TFTP Files
1:39AM 0 mediatrix 1102
1:05AM 2 Configs for Gradwell and inWeb
12:44AM 5 Extensions base policy
 
Thursday March 9 2006
TimeRepliesSubject
11:08PM 0 PRI/T-1
10:30PM 2 Asterisk Re-invites - how to tell ?
9:50PM 2 How to assign channels for asterisk
8:32PM 4 OT: Snom 320, displaying text on the scree n from *
8:22PM 0 Not getting mails from Mar 2
8:04PM 4 Sangoma A200 error
7:21PM 0 Mitel SX-2000 <--> TE210P Red Alarm
4:37PM 3 OT: Snom 320, displaying text on the screen from *
3:15PM 3 Extracting info from the $EXTEN variable
3:05PM 1 Polycom 4000 results?
2:43PM 0 Nortel BCM and Asterisk as SIP Extension
2:30PM 2 SIP/Video client for PocketPC that works with Asterisk?
2:00PM 10 IVR woes
1:50PM 1 Single E1 with HW Echo Can?
1:40PM 0 best pre paid for astreisk?
12:37PM 1 G729, G729 annex A or G729 annex B?
12:19PM 2 news-reading question
11:55AM 1 Chinaroby VOIP phones? SECOND TIME!
11:20AM 0 T1 card Ali
10:46AM 0 AMD64 x2 and asterisk 1.2.4 not hearing demo-congrats
10:27AM 2 RES: DTFM or FSK
10:16AM 1 Oneway voice
10:01AM 0 res_musiconhold.c: Only wrote -1 of 640 bytes to pipe // no queue music
9:16AM 1 SPA3000 and callerID
9:06AM 0 broken pipe, restart asterisk
8:59AM 3 TDM11B Hang up detection not working in France ?
8:46AM 2 Merlin Magix Integration
8:20AM 1 Getting to the last "old" voicemail message
8:09AM 3 DTFM or FSK
8:04AM 3 cdr data
7:44AM 0 Stress Tests from AsteriskGur with Asterisk@Home
7:39AM 0 paly sound when we Start and stop recording
7:26AM 1 Jitter buffer for SIP channels (OT?)
7:05AM 16 7940/60 SIP 8.2
7:00AM 1 Is extension.conf documentation wrong?
6:49AM 1 digium certification for Europe
6:28AM 0 ruby-agi-1.1.2 released
6:17AM 0 Re: [asterisk-biz] Professional Recordings
6:00AM 0 Music On Hold playback
5:23AM 0 Fax behind ATA
4:43AM 7 Festival tts
3:17AM 1 Asterisk code help
2:00AM 0 Attended transfer returns invalid extension
1:46AM 0 Can't hear busy tone
12:57AM 0 Does Atcom AU-200 work with XLite?
 
Wednesday March 8 2006
TimeRepliesSubject
10:53PM 0 re: Billing Package for Asterisk
8:00PM 0 Openline4 and asterisk@home
5:36PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 03/11/2006
4:45PM 0 2-Asterisk@Home Servers Connecting Portugal to Brazil (offices)
4:22PM 0 overlap dialing with polycom?
4:19PM 10 Professional Recordings
4:17PM 0 Unicall, Fax and Echo cancellation
4:05PM 1 Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???
3:42PM 0 Random Zap port going crazy When channel released after a flash.
2:30PM 1 Any way to change dns timeout value? Asterisk hangs if internet unreachable
2:14PM 0 List Problems
1:49PM 0 More 7940 Questions
1:43PM 13 Cisco 7960 SIP - Displaying Time
12:32PM 1 Zap not installing
12:17PM 0 Faxing with MFC/r2
12:16PM 0 Softphone for Windows CE 3.0
12:07PM 3 No DTMF
11:48AM 3 Memory Problems
11:12AM 1 impact of qualify=yes
10:53AM 3 Putting caller in queue and dialing an extension simultaneously
10:51AM 0 RES: Inserting access codes as prefixes to CID
10:51AM 1 What port mpg123 uses for MoH?
10:47AM 1 Upgrading Asterisk witk G729 license installed
10:43AM 4 RES: pap2 Dial plan
10:43AM 6 PAP2 won't make two g729 calls at the same time
10:23AM 1 Location of MeetMe Recordings
9:54AM 0 pickup last ringing phone
9:26AM 0 Cisco Call Manager SIP trunk + Asterisk
9:14AM 4 Is everyone getting mails except me?
9:09AM 0 Chinaroby VOIP phones?
8:20AM 0 status on jitter buffer for SIP/RTP? (OT?)
8:19AM 3 Asterisk @ Home 2.6 Call hangs up
7:38AM 1 Asterisk sip and radius authentication
7:19AM 2 parking slot lights - testers wanted
6:26AM 3 [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?
6:19AM 0 Mitel SX-2000 and Asterisk integration
5:11AM 2 Calls forwarding to numbers only in user's context
5:00AM 0 Clock is runing too fast, Asterisk@home2.5 Ztdummy and VMware workstation
4:22AM 0 Conference room owner Changing his room password? Ast@Home
4:15AM 0 Size'ing/performance
3:47AM 0 can't call some numbers/providers , ani code missing in sip header (found the problem, not the solution)
3:17AM 2 REGISTER headers changed
1:15AM 4 sending text to display of sip phones
12:49AM 0 Hangup with error
 
Tuesday March 7 2006
TimeRepliesSubject
11:52PM 0 Called number not recognised
10:46PM 0 icmp 36: 192.168.30.32 udp port 5004 unreachable
10:16PM 2 MeetMe 'i' option not working correctly?
8:02PM 1 OT: Polycom Registration Weirdness
7:42PM 0 Mitel SIP firmware
7:03PM 4 Reverse group in zapata.conf
5:54PM 0 HOWTO volume per (7960) phone
5:34PM 0 IAXy (S101) echo?
5:13PM 0 Agents and agent counts
4:26PM 7 System Design
4:20PM 2 OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3
3:58PM 3 Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?
2:37PM 0 Asterisk Compatible Server Architecture
2:36PM 8 MWI, SER and asterisk
1:08PM 10 can i get the script
12:53PM 1 Question from a newbie on finding digium hosts
11:34AM 1 Changing REINVITE status of the channel dynamically
11:33AM 5 Receiving Multiple calls on asterisk at home
11:22AM 1 Call Path Optimization?
11:21AM 1 Setting Vaaibles
10:56AM 2 Using softphone from a remote location to get into *
10:53AM 5 OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 WARNING!!!
10:46AM 2 pap2 Dial plan
10:39AM 0 anonymous caller id causes crash
9:53AM 10 res_mysql.conf & DNS SRV lookup
9:31AM 1 I can't receive multiple pages with spandsp
9:28AM 1 PLEASE HELP ,a2billing problem with call duration
9:12AM 14 Oh this is bad.... bindaddr and rtp traffic
9:00AM 3 indications & SIP
8:58AM 0 Toll free nos
8:49AM 0 Destroying a SIP extension doesn't destroyvoicemail box?is this a bug?
8:15AM 1 Asterisk + SE Linux
8:02AM 2 PBX-VPN-SIP-Asterisk trouble
8:02AM 0 Destroying a SIP extension doesn't destroy voicemail box?is this a bug?
7:37AM 2 Send One Touch Record to mail
7:32AM 0 a2billing problem with call duration
6:57AM 5 Problem ChanSpy
6:40AM 3 How to change Budgetone dialtone?
6:16AM 0 webvmail problems
4:41AM 0 Two Asterisk server
4:08AM 1 Help! Connecting two Astersik via SIP channels
3:01AM 2 Asterisk Prepaid Card
3:01AM 8 ON DEMAND call Recording
2:20AM 2 Periodic-announce in queues
1:40AM 0 Gmane - Asterisk Users Mailing List
1:25AM 0 Asterisk add-ons - H323
1:11AM 0 R: Capturing DTMF during a call
 
Monday March 6 2006
TimeRepliesSubject
11:32PM 3 What is asterisk
10:29PM 1 Hangup issues
8:20PM 7 Asterisk download file locations
7:15PM 2 Confusion about construction of RURIs from contact headers for BYEs generated by *
6:45PM 2 ENUM lookup issues with e164.org
6:18PM 0 streaming recordings
5:19PM 2 PRI CID signalling not working?
4:04PM 2 Comedian Mail Add-ons?
3:57PM 1 most common VOIP echo simulaton for research purposes ?
3:39PM 1 IPv6
3:32PM 0 Asterisk-addons 1.2.2 released
2:25PM 4 Problem getting two x200p cards working on 1.2.4
2:00PM 4 call manager integration
1:43PM 0 Music on hold volume too high - using built in music on hold.
1:00PM 1 Asterisk and CISCO 7970 color
12:16PM 0 No ring when doing blind transfer.
11:34AM 1 Upgrading AAH
11:24AM 0 Initiate and monitor multiple calls?
11:11AM 1 PLEASE respond: how to get Asterisk to change coders on RTP handoff??
10:55AM 0 call files and cdr I need src different from CallerID(number)
10:53AM 4 cdr records on transfer
10:53AM 6 Polycom voice.gain.tx.analog.handset and asterisk echo
10:40AM 0 ring noise at the background
10:09AM 0 Question: "When i Diall a group"
10:02AM 0 Ringduration problem when calling out via Sip
9:45AM 1 Buddy watch?
9:44AM 1 Bad Meetme() Bug
9:44AM 0 chan_zap.c:6570 handle_init_event error
9:30AM 2 Asterisk on MacOS?
9:18AM 16 NEWS: SIP Firmware Available for Cisco 7970
9:09AM 0 Information to program a new driver for Asterisk
8:54AM 6 One Extension - Two Calls?
8:33AM 0 Set(LANGUAGE()=language) - for queue
8:08AM 8 spa3000 asterisk fxo gateway
7:46AM 2 grandstream handytone 286 sometimes dials out wrong number
7:18AM 2 Unable to make hints function properly
6:28AM 0 Outbound Proxy Support
6:12AM 0 hangup on silence?
5:31AM 1 Capturing DTMF during a call
4:28AM 1 Extension 's' in Realtime
4:26AM 0 problems in changing Festival's Default Voice in Asterisk
3:48AM 3 Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1
3:15AM 4 Two asterisks on one machine
1:21AM 0 need to find an asterisk user from Costa Rica.
1:15AM 0 Passing Digits between ISDN PBX and Asterisk
1:03AM 0 No new mails
12:42AM 1 Redirecting to another service/server
 
Sunday March 5 2006
TimeRepliesSubject
8:10PM 0 static kernel
7:37PM 0 Dial() cmd executing Macro - dropped audio
7:19PM 5 Snom 360 Hinting tricks
5:23PM 17 Polycom 501 power over ethernet
4:40PM 0 ZapATA channels up, but calls cannot be made
3:27PM 0 Sipura SPA-3000 in Egypt
1:20PM 2 Problem with libpri?
12:19PM 0 RE: Asterisk-Users Digest, Vol 20, Issue 31
10:44AM 0 re: Sixtel Services
10:27AM 0 to configure asterisk to work with the nathelper module of openser
9:28AM 0 re: Sixtel Services
8:57AM 4 Dialplan - strip IDD prefix and insert another
8:57AM 3 Inserting access codes as prefixes to CID
6:10AM 2 uniqueid
6:06AM 0 Realtime Content on LCD Display
3:27AM 2 20 seconds til voice transmission starts
3:11AM 2 Can log into the mailbox from Soft-phone , but not from Hardware Phone
 
Saturday March 4 2006
TimeRepliesSubject
6:37PM 2 Auto dial feature
2:26PM 24 Problem compiling ztdummy on centos 4, 2.6 kernel
2:12PM 2 Upgrading to 1.2.5?
1:20PM 1 # (send immediately) and dialplan broken on PAP2?
9:08AM 0 RE: Asterisk-Users Digest, Vol 20, Issue 20
8:55AM 0 (no subject)
8:54AM 2 Asterisk 1.2.5 Released
7:46AM 1 Asterisk to a Huawei softX3000
7:29AM 1 *** Yet another boring weekend? Test new Asterisk features in development!
6:04AM 2 help with asterisk installation
2:37AM 0 Accept Unregistered GK Calls
2:35AM 0 asterisk 1.2.5 cannot call a zap channel extension
2:04AM 3 What hardware to use for ISDN in Romania
12:34AM 0 Call Waiting? Should this just work?
 
Friday March 3 2006
TimeRepliesSubject
10:21PM 15 really need help with outgoing calls..PSTN errors
9:54PM 3 No audio on PRI.
6:02PM 27 MultiBRI in Australia - found one - maybe
5:29PM 2 Background() App From AGI
5:18PM 0 ANI configuration
4:17PM 0 Can we replace existing SIP call with new one?
4:02PM 2 Does an entry in AstDB stay after reboot?
3:34PM 0 Multi node call center
3:29PM 1 Meetme Timing Interface
3:24PM 4 Meetme Participant Announcement
3:19PM 1 Call Transfer - "Both legs must reside on Asterisk box to transfer at this time"
3:16PM 0 a=fmtp:18 annexb=no
2:57PM 1 Polycom 501 and single call only using AAH 2.2
2:32PM 2 IAX to SIP conversion: SIP From header issue
2:22PM 13 new beta Grandstream firmware HT488_496_386
2:03PM 0 Spontaneous reloads
2:01PM 0 CON-SNT-CP7970 resellers?
1:06PM 18 Preferred editor(s) dialplan coding?
1:01PM 0 Problem with HT-286 & BT-101
12:56PM 4 Two PBX
12:34PM 0 [Fwd: Re: problem with incoming peer (cisco as5400)]
12:14PM 0 Bad quality between SIP and TDM
12:10PM 0 Asterisk coder conflicts
11:46AM 1 SIP Problem - Asterisk to Provider Gateway
10:57AM 1 IAX2 register problem
10:49AM 2 Hardware Requirements for 1M minutes
10:48AM 0 [HELP] dial plan continue for outbound channel on disconnect
10:48AM 2 dtmf tones problem with unicall and E1
9:26AM 0 sprint FNTM(sp?) line
9:19AM 7 Echo Cancelation on TE110P
9:17AM 0 misdn <--> zap problem
9:04AM 2 what version s this??
9:03AM 8 Sipura RMA
8:37AM 0 Fw: 2 real phone numbers on one SIP account
8:32AM 17 Program Buttons on Cisco 79xx Phones
8:07AM 0 Asterisk Realtime voicemail question
8:06AM 2 Autofill phonebook??
8:02AM 0 Realtime Extensions hint priority
7:36AM 0 Thinking of moving from pure VoIP to PRI - thoughts?
7:11AM 0 'quit' isn't in the CLI's 'help'
6:57AM 0 check call status during call
6:57AM 0 calls only for logging users
6:56AM 8 Asterisk Fax Question
6:31AM 11 web meetme instructions
6:09AM 0 Implementing MOH while trunks gets connected...
5:37AM 4 login/logout agents in a specific queue
3:53AM 0 Part-Time work available
3:13AM 0 is there a variable for the calling IP ?
2:45AM 1 Problem with NAT!!!
2:17AM 0 Status of another channel from AGI
 
Thursday March 2 2006
TimeRepliesSubject
11:11PM 0 Australian E1 from Optus still not working
11:10PM 4 snom 320 MWI light
7:32PM 2 TIMESTAMP, DATETIME not working
5:52PM 1 RE: [on-asterisk] Brainstorming dual-core and Asterisk
5:14PM 0 RE: [on-asterisk] containers, virtualization, and high availability
3:56PM 2 Setting Max Calls on an IAX trunk
3:40PM 8 MixMonitor Problems -- sssshh, don't be too loud
2:10PM 1 Sip Realtime Configs Samples with MySQL
1:34PM 0 * dials out zap line first 6 digits, pause, then last digit
1:19PM 6 Child PID's
1:17PM 11 Changing caller id on transfer
1:13PM 0 Frequently Showed Info Messages
12:17PM 0 RE: Asterisk-Users Digest, Vol 20, Issue 13
12:14PM 7 G729 and Meetme
11:46AM 0 OT - Cisco IP Phone and PC in different VLANs(with802.1x)
11:44AM 0 OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x)
11:24AM 1 setmusiconhold doesn't work between 2 SIP phones
10:46AM 20 Polling Asterisk for Life
10:42AM 3 wake up calls
10:27AM 0 [HELP] Outbound Channel next priority on originator disconnect
9:58AM 1 Toshiba DK424 / Asterisk / DTMF problems
9:51AM 2 [Fwd: Over 40 destinations for FREE!]
9:41AM 0 problems with MOH
9:16AM 0 List disabled notification
9:02AM 0 problem with incoming peer (cisco as5400)
8:51AM 0 Native attended transfer: taking again original conversation
8:46AM 0 Redirect a sip outbound requests to a sip proxy
8:02AM 3 Get no busy signal on my analog line
8:01AM 0 channels appear to be stuck
7:28AM 0 Parked calls delay
6:53AM 0 gotoiftime with list of time range
6:46AM 2 test call quality
6:43AM 4 Sipura SPA-3000 vs Linksys SPA3000
6:23AM 1 IAX Video and Meetme
6:01AM 0 remote IP address in channel?
5:44AM 4 error messages on /var/log/asterisk/messages
4:30AM 3 Native music on hold - Error
3:54AM 1 dial plan !!
3:50AM 5 Milliwatt Analyzer available
3:01AM 0 TE40X zapata.conf configuration sample
2:46AM 1 Managed Switches QoS to deal with network bottleneck
1:55AM 0 HDLC error
12:59AM 31 asterisk management interface
12:45AM 6 Info about F1000G
 
Wednesday March 1 2006
TimeRepliesSubject
11:36PM 0 hands-on experience with soft videophone
7:37PM 0 Want to record call and put into users voicemail
6:56PM 12 Polycom 501
5:41PM 1 Settings for Yuxin Phones...
5:32PM 5 TE411P VPM
4:16PM 4 Two FXOs getting bridged?
2:40PM 0 queues & tranfers
1:56PM 2 GXP-2000 Volume Issue
1:42PM 9 my zap channel not ringing
12:58PM 0 Agents and Chanspy
11:54AM 0 perl AGI won't run from extensions.conf
11:38AM 8 OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)
11:08AM 0 Variables in queues.conf
11:04AM 2 ignore a DID?
9:59AM 4 160 analogue phones..
9:26AM 10 Lowering Server Load
8:51AM 0 SendDTMF in connected call?
7:52AM 0 Configuration call hijack for users in a hunting group ?
7:51AM 9 Asterisk transfer conflict
7:48AM 2 SIP contexts being confused
6:49AM 0 Help with Digium TE210P, TDM400P card in Dell PE830
5:59AM 0 Early media and custom SIP return codes
5:39AM 1 Agents, queues and Pentalties
5:02AM 7 Same CID on multiple users(friends9 in SIP.conf
4:58AM 9 MOH native files
4:10AM 1 how to run asterisk?
3:04AM 0 No sound after loading module wcte11xp
3:03AM 5 about operator
2:39AM 0 T38 fax pass thru to Cisco as53xx
2:11AM 0 Cisco 7905 - vad, cng
1:49AM 0 Callerid error: receiving DNID instead of callerID
1:27AM 0 ooh323 codec's - alaw
1:17AM 1 Cisco Callmanager integration with asterisk
12:38AM 2 Cannot log into mailbox , guidance requested
12:13AM 10 Working Asterisk with Austrian ISDN p2p